1. Technical Field
This invention concerns the transporting of digital media packets, such as audio and video. In a first aspect the invention concerns a data network able to carry high fidelity audio. In another aspect the invention concerns a network device for sending and receiving digital media packets; such as a Digital Signal Processing (DSP) chip for public address systems. In a further aspect the invention concerns a method of operating the data network or network devices and software for performing the method, for instance, on a personal computer.
2. Background Art
Audio and video signals have long been transmitted using application specific cables. For instance two-core speaker cable is used to carry left and right audio signals from amplifiers to speakers.
The time at which a received media signals are played out by a media device is called the playout time. Typically, a media device that receives media signals will playout the media signals by rendering them in some way. For example, if the media device is a loudspeaker it will render the audio media signal into sound. If the media device is a video screen it will render frames of the video media signal onto a screen. Alternatively, if the media device is a lighting control system it will render the lighting media signal by turning a spotlight on and off.
Real time transporting of digital audio and video and other digital media over data networks creates a new set of problems compared to non-media data. For instance, data networks may use packet switching in which data is divided into packets for separate transmission. As the packets are transmitted, sequential packets may take different routes and have different transit times. The packets are numbered to ensure they can be reordered correctly after arrival. This technique, however, does not suffice when left and right audio signals are to be received at different destinations, for instance at different speakers.
Unlike non-media data, digital media must be played out in synchronisation. For example, video and audio must be aligned in time so that when they are played out the images match the sound.
The concept of a network clock has been used to address timing problems in data networks. A network clock signal is typically generated at a specific point in the network and this becomes the system time signal received by devices on the network. The system time signal is then used as a time reference for every device that receives the system time signal. Because of the topology of the network, devices at different locations on the network will receive the clock signal with a phase offset from the network clock, depending on the propagation delay from the clock to the device. A further consequence is that the different remote devices will have received clock signals that have phase offsets with respect to each other, as well as with respect to the network clock.
Digital media transmission has historically embedded clocking information in the transmitted data. Embedding and recovering clocking information from data signal transitions or packet timing (e.g. AES3, SP/DIF, Gibson MaGIC) works well for point to point links between a small numbers of devices, but as the number of devices increases, clock jitter cascades and builds through each device that recovers and re-transmits the clocking information. Large systems employ a separate clock network to avoid such problems with clock jitter.
Digital media transmissions may alternatively employ a Time Division Multiplexing (TDM) approach. In TDM systems (e.g. MADI, CobraNet), a master clock device initiates periodic transmission cycles and each device is allocated one or more time slots within that cycle for transmission. This limits the total available number of channels.